mirror of
https://github.com/moonlight-stream/moonlight-embedded.git
synced 2026-02-16 02:20:42 +00:00
Provide alternative SDL audio output
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@@ -36,7 +36,7 @@ if(FREESCALE_FOUND)
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endif()
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if (AVCODEC_FOUND AND AVUTIL_FOUND AND SWSCALE_FOUND AND SDL_FOUND)
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list(APPEND SRC_LIST ./src/video/ffmpeg.c ./src/video/sdl.c)
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list(APPEND SRC_LIST ./src/video/ffmpeg.c ./src/video/sdl.c ./src/audio/sdl.c)
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list(APPEND MOONLIGHT_DEFINITIONS HAVE_SDL)
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endif()
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@@ -22,3 +22,6 @@
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extern const char* audio_device;
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extern AUDIO_RENDERER_CALLBACKS audio_callbacks_alsa;
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#ifdef HAVE_SDL
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extern AUDIO_RENDERER_CALLBACKS audio_callbacks_sdl;
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#endif
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79
src/audio/sdl.c
Normal file
79
src/audio/sdl.c
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@@ -0,0 +1,79 @@
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/*
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* This file is part of Moonlight Embedded.
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*
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* Copyright (C) 2015 Iwan Timmer
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*
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* Moonlight is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 3 of the License, or
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* (at your option) any later version.
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*
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* Moonlight is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with Moonlight; if not, see <http://www.gnu.org/licenses/>.
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*/
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#include "../audio.h"
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#include <SDL.h>
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#include <SDL_audio.h>
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#include <stdio.h>
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#include <opus.h>
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#define SAMPLE_RATE 48000
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#define CHANNEL_COUNT 2
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#define FRAME_SIZE 240
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static OpusDecoder* decoder;
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static short pcmBuffer[FRAME_SIZE * CHANNEL_COUNT];
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static SDL_AudioDeviceID dev;
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static void sdl_renderer_init() {
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int rc;
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decoder = opus_decoder_create(SAMPLE_RATE, CHANNEL_COUNT, &rc);
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SDL_InitSubSystem(SDL_INIT_AUDIO);
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SDL_AudioSpec want, have;
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SDL_zero(want);
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want.freq = SAMPLE_RATE;
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want.format = AUDIO_S16LSB;
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want.channels = CHANNEL_COUNT;
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want.samples = 4096;
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dev = SDL_OpenAudioDevice(NULL, 0, &want, &have, SDL_AUDIO_ALLOW_FORMAT_CHANGE);
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if (dev == 0) {
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printf("Failed to open audio: %s\n", SDL_GetError());
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} else {
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if (have.format != want.format) // we let this one thing change.
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printf("We didn't get requested audio format.\n");
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SDL_PauseAudioDevice(dev, 0); // start audio playing.
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}
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}
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static void sdl_renderer_cleanup() {
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if (decoder != NULL)
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opus_decoder_destroy(decoder);
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SDL_CloseAudioDevice(dev);
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}
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static void sdl_renderer_decode_and_play_sample(char* data, int length) {
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int decodeLen = opus_decode(decoder, data, length, pcmBuffer, FRAME_SIZE, 0);
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if (decodeLen > 0) {
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SDL_QueueAudio(dev, pcmBuffer, decodeLen * CHANNEL_COUNT * sizeof(short));
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} else {
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printf("Opus error from decode: %d\n", decodeLen);
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}
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}
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AUDIO_RENDERER_CALLBACKS audio_callbacks_sdl = {
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.init = sdl_renderer_init,
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.cleanup = sdl_renderer_cleanup,
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.decodeAndPlaySample = sdl_renderer_decode_and_play_sample,
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};
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@@ -72,6 +72,10 @@ DECODER_RENDERER_CALLBACKS* platform_get_video(enum platform system) {
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AUDIO_RENDERER_CALLBACKS* platform_get_audio(enum platform system) {
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switch (system) {
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#ifdef HAVE_SDL
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case SDL:
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return &audio_callbacks_sdl;
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#endif
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default:
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return &audio_callbacks_alsa;
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}
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